Good idea but this would cause to play songs with a high amplitude (this
are "normal" songs as the amplitude should use the full range of 16 bits)
with a low output gain, to catch up songs with a low amplitude. The output
gain of my soundcard is quiet low and I would receive more noise from
analog amplification.

However normalization will be implemented, it must be a addable feature,
not standard behavior. Sometimes it is necessary to have unmodified
amplitude. For example CDs without gaps between songs. A jump in volume
would result. I use a self written tool to normalize the samples by first
scanning through the file, then adding a constant value to all samples
lifting maximum peak to boundaries. It takes quiet a lot of time (30 sec
on 30Megs on K6-200, just a guess, am on different computer now).








Greg Maxwell <[EMAIL PROTECTED]> on 17.01.2000 21:10:35

Please respond to [EMAIL PROTECTED]
                                                                                
                                                                                
                                                                                


                                                              
                                                              
                                                              
 To:      [EMAIL PROTECTED]                              
                                                              
 cc:      (bcc: Thomas Marschall/MUC/Santix)                  
                                                              
                                                              
                                                              
 Subject: Re: [MP3 ENCODER] Normalization routine?            
                                                              







On Mon, 17 Jan 2000 [EMAIL PROTECTED] wrote:

> I think performance loss due to rounding errors can be ignored because:
[snip]

Insted of normalizaing your mp3 collection while you encode it,
potentially losing some (even a small amount) of quality/dynamic range,
why not just add compute a 'perceptual volume index' while you encode, and
stick it on the file ALA id3, the decoder would then adjust your mixer
levels between files to keep them closer togeather.


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